Re: [asterisk-users] [asterisk-app-dev] Handling transfers with ARI
Thank you all for the hints.
I ended up using a mix of dialplan to deal with the Local channels, and
ARI to detect the transfer and redirect. It doesn't look like a "clean"
solution but I have nothing better for the moment :
Dialplan :
exten =
100,1,GotoIf($[$["${SIPTRANSFER}"="yes"]&$["${CHANNEL(channeltype)}"="Local"]]?waittransfer:)
; deal with channel being transfered
same = n,Transfer(100)
same = n,Hangup()
; deal with Asterisk-managed Local channel
same = n(waittransfer),Wait(2)
same = n,Hangup()
NodeJS :
bridge.once('BridgeAttendedTransfer', event => {
var transferee = new ari.Channel(event.transferee.id);
transferee.continueInDialplan({
context: event.context,
extension: event.exten,
priority: 1
});
});
Le 23/12/2020 à 19:46, Phil Mickelson a écrit :
Unfortunately, I suspect my situation is different from yours in that
I control everything. And, when Bob wants to transfer the call he
clicks a button on the screen, not a button on the phone. I don't use
any part of the dialplan except to start ARI.
Sorry.
Phil
On Wed, Dec 23, 2020 at 2:56 AM Jean Aunis > wrote:
Thanks for the answer.
Not sure I get the idea : when a SIP phone performs a
blind-transfer, I have no control over what Asterisk does with the
channels. During my tests, Bob's channel was automatically pulled
out of the bridge, and replaced with a Local channel whose peer
goes through the dialplan to the transfer destination.
How can you link the newly created Local channel with Alice's one ?
For the moment, I have a piece of solution with the
BridgeBlindTransfer event, but I still have troubles with these
Local channel issues.
Le 22/12/2020 à 20:13, Phil Mickelson a écrit :
Not sure if this will help but what I do is fairly simple. A
couple of things:
1. This is all written in JS using Node.js.
2. I use ari-client from npm.
To me this is very simple. You already have the bridge and
channel setup for Alice. I create another channel that dials
Charlie. And, as soon as the create channel call comes back I
just set the channel id (was Bob) in the bridge to the new
channel for Charlie. That's it. If it doesn't get answered I
hope it goes to VM. However, that's the downside of a blind
transfer. I have some code in there for what happens if Alice
hangs up before Charlie answers, etc but that's because I keep
track of every call in my system.
And I wrote all of this before there were Promises and
Async/Await. Hopefully next year I'll have the time to rewrite
the whole thing.
And, for the people at Asterisk who came up with the idea of
ARI. Thank you s much. Hope everyone has a wonderful
holiday and that 2021 is much better than 2020!
Phil
On Tue, Dec 22, 2020 at 5:38 AM Jean Aunis mailto:jean.au...@prescom.fr>> wrote:
Hello,
I'm struggling to find a way to properly handle blind
transfers with ARI.
This is my use case :
- Alice calls Bob through Asterisk
- dialing and bridging is done with ARI
- when Bob blind-transfers to Charlie, I would like to use the
"redirect" ARI operation, or the Transfer application
But here is the issue : since the channels are stasis-managed,
transferring is done with Local channels which remain in the
path, so
Transfer and redirect have no effect on them. And Alice's
channel is not
aware that it is being transferred.
Has somebody already dealt with this ?
Regards,
Jean
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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI
On Thu, 24 Dec 2020, Turritopsis Dohrnii Teo En Ming wrote: 3. secret is 8 char only, must be numeric My my SIP.cnf file from 2007 contains: image_version: P0S3-8-12-00 line1_password: 346cc89a2526255839534c22ad7790c and my notes say my 9760 only allowed up to 31 character passwords. You may find it useful to use tcpdump with '-w' to write the packets to a file and then analyze with sngrep. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI
Hi Turritopsis I think the key point maybe making sure the password doesn’t exceed the capacity of the phone. So an 8 char password is a good idea I would be surprised if pjsip doesn’t work but I haven’t tried it with a Cisco phone Whatever gets you working is what you want Have a wonderful Xmas Cheers Duncan > On 24/12/2020, at 1:11 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Thank you for your replies, Duncan Turnbull. > > I am going to run tcpdump on my Asterisk PBX server. > > By the way, I found a Youtube video. > > Youtube video: Cisco 7942g IP Phone Configuration on FreePBX In-Depth(Without > Endpoint Manager) > > Link: https://www.youtube.com/watch?v=gk6w8O3fZlc&feature=youtu.be > > From the above youtube video, it seems that I cannot use pjsip extension for > my Cisco 7960 IP phone. I need to delete the pjsip extension, and then create > a legacy chan_sip extension, it seems. > > These are the notes I have taken after watching the above Youtube video: > > 1. Cannot use pjsip extension, need to use legacy chan_sip extension > > 2. Display name: Your name > > 3. secret is 8 char only, must be numeric > > 4. Voicemail: Enabled > > 5. Require from same extension: yes > > 6. Go to Advanced, nat mode: never > > 7. Port 5060 > > 8. Qualify: No > > 9. Send RPID: Send Remote-Party-ID header > > 10. Go to Settings > Asterisk SIP Settings > SIP Legacy Settings (chan_sip) > > 11. NAT: No > > 12. Enable SRV Lookup: No > > 13. Edit SEP.cnf.xml, sipPort: 5160 > > 14. Line #1, port: 5160 > > > >> On 2020-12-23 17:55, Duncan Turnbull wrote: >> >> Sent from my iPad On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming wrote: >>> Hi Duncan Turnbull, >>> You can watch my Youtube video of my Cisco 7960 IP phone. >>> The link is: https://www.youtube.com/watch?v=ip_F08jmmio >>> My Youtube video shows the Network Configuration settings, SIP >>> Configuration settings and Status of my Cisco 7960 IP Phone. >> The phone looks like it has picked up the configs however in the >> status there are two error messages re parsing SipDefault.cnf and the >> specific SIP..MAC.. file - you should try and remedy those errors . >> Otherwise most of the settings look to be there >> I would suggest cutting out as much of the config as you can >> I would also suggest you run tcpdump on the 192.168.1.9 box and >> monitor any traffic at all coming from your phone which is now on >> 192.168.1.130. You may see the SIP messages there >> Cheers Duncan >>> Did you see anything wrong? On 2020-12-23 12:38, Duncan Turnbull wrote: Hi there >> On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming >> wrote: > Good morning Duncan Turnbull, > I have posted my Asterisk PBX server debugging output previously in my > original post. The link is: > http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html > I saw many REGISTER requests. Are these REGISTER requests from my Cisco > 7960 IP phone? Could you help me to check? Thank you very much. If they come from the phone they will have the phones ip address. The phone will also try and register with the extension you have given it. None of the registration messages appear to have the up or the extension so you will need to figure out what’s gone wrong with the phones config That’s why checking the phone settings to see fit they have changed helps understand if your configs were correct. You can do this via the phone screen or telnet. It will take you some time to become familiar with this but it’s worth it Good luck > I shall reproduce my Asterisk PBX server debugging output below. > SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT > = > # asterisk -vvvr > sip set debug on > freepbx*CLI> > [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister: > -- Re-registration for 60...@sip.sg.didlogic.net > REGISTER 12 headers, 0 lines > Reliably Transmitting (NAT) to 107.6.123.181:5060: > REGISTER sip:sip.sg.didlogic.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport > Max-Forwards: 70 > From: ;tag=as6df6d977 > To: > Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1 > CSeq: 165 REGISTER > Supported: replaces, timer > User-Agent: FPBX-15.0.16.81(16.13.0) > Authorization: Digest username="60751", realm="sip.sg.didlogic.net", > algorithm=MD5, uri="sip:sip.sg.didlogic.net", > nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=", > response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth, > cnonce="2b1b6d13", nc=0003 > Expires: 120 > Contact: > Content-Length: 0 > --- > <--- SIP read from UDP:107.6.123.181:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received= OFFICE PUBLIC IP>
[asterisk-users] asterisk-13 MWI - phone not blinking
On asterisk-11 MWI was working correctly, phone message light was blinking (standard phone). With asterisk-13, this feature is not working. Who to trouble shoot? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI
Thank you for your replies, Duncan Turnbull. I am going to run tcpdump on my Asterisk PBX server. By the way, I found a Youtube video. Youtube video: Cisco 7942g IP Phone Configuration on FreePBX In-Depth(Without Endpoint Manager) Link: https://www.youtube.com/watch?v=gk6w8O3fZlc&feature=youtu.be From the above youtube video, it seems that I cannot use pjsip extension for my Cisco 7960 IP phone. I need to delete the pjsip extension, and then create a legacy chan_sip extension, it seems. These are the notes I have taken after watching the above Youtube video: 1. Cannot use pjsip extension, need to use legacy chan_sip extension 2. Display name: Your name 3. secret is 8 char only, must be numeric 4. Voicemail: Enabled 5. Require from same extension: yes 6. Go to Advanced, nat mode: never 7. Port 5060 8. Qualify: No 9. Send RPID: Send Remote-Party-ID header 10. Go to Settings > Asterisk SIP Settings > SIP Legacy Settings (chan_sip) 11. NAT: No 12. Enable SRV Lookup: No 13. Edit SEP.cnf.xml, sipPort: 5160 14. Line #1, port: 5160 On 2020-12-23 17:55, Duncan Turnbull wrote: Sent from my iPad On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming wrote: Hi Duncan Turnbull, You can watch my Youtube video of my Cisco 7960 IP phone. The link is: https://www.youtube.com/watch?v=ip_F08jmmio My Youtube video shows the Network Configuration settings, SIP Configuration settings and Status of my Cisco 7960 IP Phone. The phone looks like it has picked up the configs however in the status there are two error messages re parsing SipDefault.cnf and the specific SIP..MAC.. file - you should try and remedy those errors . Otherwise most of the settings look to be there I would suggest cutting out as much of the config as you can I would also suggest you run tcpdump on the 192.168.1.9 box and monitor any traffic at all coming from your phone which is now on 192.168.1.130. You may see the SIP messages there Cheers Duncan Did you see anything wrong? On 2020-12-23 12:38, Duncan Turnbull wrote: Hi there On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming wrote: Good morning Duncan Turnbull, I have posted my Asterisk PBX server debugging output previously in my original post. The link is: http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html I saw many REGISTER requests. Are these REGISTER requests from my Cisco 7960 IP phone? Could you help me to check? Thank you very much. If they come from the phone they will have the phones ip address. The phone will also try and register with the extension you have given it. None of the registration messages appear to have the up or the extension so you will need to figure out what’s gone wrong with the phones config That’s why checking the phone settings to see fit they have changed helps understand if your configs were correct. You can do this via the phone screen or telnet. It will take you some time to become familiar with this but it’s worth it Good luck I shall reproduce my Asterisk PBX server debugging output below. SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT = # asterisk -vvvr sip set debug on freepbx*CLI> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister: -- Re-registration for 60...@sip.sg.didlogic.net REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 107.6.123.181:5060: REGISTER sip:sip.sg.didlogic.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport Max-Forwards: 70 From: ;tag=as6df6d977 To: Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1 CSeq: 165 REGISTER Supported: replaces, timer User-Agent: FPBX-15.0.16.81(16.13.0) Authorization: Digest username="60751", realm="sip.sg.didlogic.net", algorithm=MD5, uri="sip:sip.sg.didlogic.net", nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=", response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth, cnonce="2b1b6d13", nc=0003 Expires: 120 Contact: Content-Length: 0 --- <--- SIP read from UDP:107.6.123.181:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received= From: ;tag=as6df6d977 To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21 Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1 CSeq: 165 REGISTER Contact: IP>:26462>;expires=120;received="sip:IP>:26462" Content-Length: 0 <-> --- (8 headers 0 lines) --- [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961 handle_response_register: Outbound Registration: Expiry for sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s) Really destroying SIP dialog '005dbc8238e06ac421ef613a3b55e134@127.0.0.1' Method: REGISTER <--- SIP read from UDP:107.6.123.181:5060 ---> <-> Reliably Transmitting (NAT) to 107.6.123.181:5060: OPTIONS sip:sip.sg.didlogic.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport Max-Forwards: 70 From: "Unknown" ;tag=as41ddf4a6 To: Contact: Call-ID: 0b0605df4f4ca7b03402c9fd5a869
[asterisk-users] Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded
FOSDEM - Real Time Communications devroom CfP = NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time slot. Overview [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2021 takes place 6-7 February 2021 and for the very first time, it will be online. This document contains information about: - Real-Time Communications developer room (devroom) - speaking opportunities - volunteering New rules for the online edition This year FOSDEM will be fully online instead of being held in Brussels, here are the most important things to know about this (quite significant) change: - The reference time will be Brussels local time (CET) - Talks will be pre-recorded in advance, and streamed during the event - The Q/A session will be live - A facility will be provided for people watching to chat between themselves - A facility will be provided for people watching to submit questions Call for participation - Real Time Communications (RTC) --- The Real-Time devroom is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers who can help manage the scheduling and live Q&A sessions.** The devroom is only on Saturday, 6th of February 2021. To discuss the devroom, volunteer or ask questions, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Key dates - 20th Dec: Submission deadline (extended to 8 January) - 24th Dec: Announcement of selected talks - 15th Jan: Presentations upload deadline - 6th & 7th Feb: Conference dates (online) - 6th Feb: RTC devroom date (online) ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 20th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM21/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. This year this is more true than ever, being able to record your presentation offline without an audience in front can greatly help build up one's confidence! ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. ### Recording help The devroom organization is able to provide help with recording your session. The recording would be performed at a scheduled time with one of us, so you won't be alone giving your presentation. Minimal edits will be possible, but the ideal plan is to record it in one shot. Thanks Dan Jenkins for providing us with the means to do this! Volunteers needed - To make the devroom run successfully, we are looking for volunteers. This year many things be done for the first time, so all the help we can get is more than welcome. Spread the word and discuss --- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to t
[asterisk-users] db_execute_sql: Error executing SQL (COMMIT): database is locked
In: /var/lib/asterisk -rw-r--r-- 1 asterisk asterisk 12288 Dec 23 10:52 astdb.sqlite3 ast_db_put: Couldn't execute statement: SQL logic error ast_db_put: Couldn't execute statement: attempt to write a readonly database db_execute_sql: Error executing SQL (COMMIT): database is locked The astdb.sqlite3 was root:root but I change it to asterisk:asterisk and I still getting these error messages -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel
On 12/23/2020 09:54 AM, Doug Lytle wrote: > Review your features.conf file in /etc/asterisk > > Doug I found id. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel
Review your features.conf file in /etc/asterisk Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel
I just upgraded to asterisk-13 (from 11) and I get some errors: 1.) Unknown DYNAMIC_FEATURES item 'automon' on channel SIP Unknown DYNAMIC_FEATURES item 'automon' on channel IAX2/voip Does anybody know how to get rid of them? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing MP3's in Asterisk
Thank you, I'm using ControlPlayback, Playback, Background etc from dialplan and AGI ... not MoH. All the files are remote so no conversion possible. Thanks On Wed, 23 Dec 2020 at 12:47, Dovid Bender wrote: > John, > > In what application are you using mp3's? In my case it was for music on > hold. I simply had asterisk call ffmpeg to play the files (to get around > all the issues). Where I have local files I convert them over to wav or gsm > and call it a day. > > > On Wed, Dec 23, 2020 at 4:33 AM Jonathan H wrote: > >> Hi all, >> >> Returning to the issue of mp3 support in Asterisk, it seems it is using a >> build from 1997?! >> http://svn.digium.com/svn/thirdparty/mp3/trunk/layer3.c >> >> I have the same problems as everyone with the mp3 add-on, but now a new >> one: >> >>- "mp3/interface.c: Junk at the beginning of frame 49443304" for ALL >>mp3 files, even when debug/verbose is off. >>- Very quiet playback compared to slin, so I have to control the >>outgoing volume by +7 for mp3 playback. >>- Now I find it chokes on mp3s which have a png cover art - this is >>perfectly valid in mp3 id3 >> >> But I still need to use ControlPlayback from dialplan and f-agi. >> >> I started going down a rabbit-hole of ARI media and Audiosocket etc, for >> example: >> >> https://www.youtube.com/watch?v=tjduXbZZEgI >> https://github.com/CyCoreSystems/audiosocket >> >> These all seem to add a fair amount of complexity for controlling >> playback - for example: >> >> https://wiki.asterisk.org/wiki/display/AST/ARI+and+Media%3A+Part+2+-+Playbacks#ARIandMedia:Part2-Playbacks-Controllingplaybacks >> And I need to know the current cplaybackoffset at any given time. >> >> I was thinking: to keep it simple, is there a way to tell Asterisk to use >> a different mpg123 (the system one - my currently installed version from >> 2020!) instead of the Asterisk build? >> >> I can live with the "junk" warnings and the quiet playback, but it cannot >> be crashing on valid id3 content (and I have no control over the remotely >> hosting files). >> >> Thanks for any pointers in the right direction! >> >> On Tue, 3 Sept 2019 at 16:20, Dovid Bender wrote: >> >>> Ludovic, >>> >>> Currently using Asterisk 14.5.0 with >>> [root@a1-c2-nyc3 ~]# rpm -qa | grep mpg123 >>> mpg123-1.25.6-1.el7.x86_64 >>> mpg123-libs-1.25.6-1.el7.x86_64 >>> mpg123-devel-1.25.6-1.el7.x86_64 >>> [root@a1-c2-nyc3 ~]# >>> Should I be using a newer version of mpg123? >>> >>> >>> >>> On Tue, Sep 3, 2019 at 11:11 AM wrote: >>> Hi ! Which Asterisk version do you use? Asterisk uses the mpg123 library. Regards, Ludovic On Sun, Sep 01, 2019 at 03:27:59PM -0400, Dovid Bender wrote: >Hi, >Does anyone know what code (if any)Â format_mp3 uses? I am trying to play >a MP3 (e.g. -- https://drolez.com/blog- Marketing automation and Web dev https://chezsandro.com - A cool place in Cape Verde :) https://aopensource.com- Android Open Source Portal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/m
Re: [asterisk-users] Playing MP3's in Asterisk
John, In what application are you using mp3's? In my case it was for music on hold. I simply had asterisk call ffmpeg to play the files (to get around all the issues). Where I have local files I convert them over to wav or gsm and call it a day. On Wed, Dec 23, 2020 at 4:33 AM Jonathan H wrote: > Hi all, > > Returning to the issue of mp3 support in Asterisk, it seems it is using a > build from 1997?! > http://svn.digium.com/svn/thirdparty/mp3/trunk/layer3.c > > I have the same problems as everyone with the mp3 add-on, but now a new > one: > >- "mp3/interface.c: Junk at the beginning of frame 49443304" for ALL >mp3 files, even when debug/verbose is off. >- Very quiet playback compared to slin, so I have to control the >outgoing volume by +7 for mp3 playback. >- Now I find it chokes on mp3s which have a png cover art - this is >perfectly valid in mp3 id3 > > But I still need to use ControlPlayback from dialplan and f-agi. > > I started going down a rabbit-hole of ARI media and Audiosocket etc, for > example: > > https://www.youtube.com/watch?v=tjduXbZZEgI > https://github.com/CyCoreSystems/audiosocket > > These all seem to add a fair amount of complexity for controlling playback > - for example: > > https://wiki.asterisk.org/wiki/display/AST/ARI+and+Media%3A+Part+2+-+Playbacks#ARIandMedia:Part2-Playbacks-Controllingplaybacks > And I need to know the current cplaybackoffset at any given time. > > I was thinking: to keep it simple, is there a way to tell Asterisk to use > a different mpg123 (the system one - my currently installed version from > 2020!) instead of the Asterisk build? > > I can live with the "junk" warnings and the quiet playback, but it cannot > be crashing on valid id3 content (and I have no control over the remotely > hosting files). > > Thanks for any pointers in the right direction! > > On Tue, 3 Sept 2019 at 16:20, Dovid Bender wrote: > >> Ludovic, >> >> Currently using Asterisk 14.5.0 with >> [root@a1-c2-nyc3 ~]# rpm -qa | grep mpg123 >> mpg123-1.25.6-1.el7.x86_64 >> mpg123-libs-1.25.6-1.el7.x86_64 >> mpg123-devel-1.25.6-1.el7.x86_64 >> [root@a1-c2-nyc3 ~]# >> Should I be using a newer version of mpg123? >> >> >> >> On Tue, Sep 3, 2019 at 11:11 AM wrote: >> >>> Hi ! >>> >>> Which Asterisk version do you use? >>> Asterisk uses the mpg123 library. >>> >>> Regards, >>> >>> Ludovic >>> >>> On Sun, Sep 01, 2019 at 03:27:59PM -0400, Dovid Bender wrote: >>> >Hi, >>> >Does anyone know what code (if any)Â format_mp3 uses? I am trying >>> to play >>> >a MP3 (e.g. >>> >>> -- >>> https://drolez.com/blog- Marketing automation and Web dev >>> https://chezsandro.com - A cool place in Cape Verde :) >>> https://aopensource.com- Android Open Source Portal >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI
Sent from my iPad > On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > You can watch my Youtube video of my Cisco 7960 IP phone. > > The link is: https://www.youtube.com/watch?v=ip_F08jmmio > > My Youtube video shows the Network Configuration settings, SIP Configuration > settings and Status of my Cisco 7960 IP Phone. The phone looks like it has picked up the configs however in the status there are two error messages re parsing SipDefault.cnf and the specific SIP..MAC.. file - you should try and remedy those errors . Otherwise most of the settings look to be there I would suggest cutting out as much of the config as you can I would also suggest you run tcpdump on the 192.168.1.9 box and monitor any traffic at all coming from your phone which is now on 192.168.1.130. You may see the SIP messages there Cheers Duncan > Did you see anything wrong? > > > > >> On 2020-12-23 12:38, Duncan Turnbull wrote: >> Hi there On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming wrote: >>> Good morning Duncan Turnbull, >>> I have posted my Asterisk PBX server debugging output previously in my >>> original post. The link is: >>> http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html >>> I saw many REGISTER requests. Are these REGISTER requests from my Cisco >>> 7960 IP phone? Could you help me to check? Thank you very much. >> If they come from the phone they will have the phones ip address. The >> phone will also try and register with the extension you have given it. >> None of the registration messages appear to have the up or the >> extension so you will need to figure out what’s gone wrong with the >> phones config >> That’s why checking the phone settings to see fit they have changed >> helps understand if your configs were correct. You can do this via the >> phone screen or telnet. It will take you some time to become familiar >> with this but it’s worth it >> Good luck >>> I shall reproduce my Asterisk PBX server debugging output below. >>> SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT >>> = >>> # asterisk -vvvr >>> sip set debug on >>> freepbx*CLI> >>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister: >>> -- Re-registration for 60...@sip.sg.didlogic.net >>> REGISTER 12 headers, 0 lines >>> Reliably Transmitting (NAT) to 107.6.123.181:5060: >>> REGISTER sip:sip.sg.didlogic.net SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport >>> Max-Forwards: 70 >>> From: ;tag=as6df6d977 >>> To: >>> Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1 >>> CSeq: 165 REGISTER >>> Supported: replaces, timer >>> User-Agent: FPBX-15.0.16.81(16.13.0) >>> Authorization: Digest username="60751", realm="sip.sg.didlogic.net", >>> algorithm=MD5, uri="sip:sip.sg.didlogic.net", >>> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=", >>> response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth, >>> cnonce="2b1b6d13", nc=0003 >>> Expires: 120 >>> Contact: >>> Content-Length: 0 >>> --- >>> <--- SIP read from UDP:107.6.123.181:5060 ---> >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP >>> 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=>> OFFICE PUBLIC IP> >>> From: ;tag=as6df6d977 >>> To: >>> ;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21 >>> Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1 >>> CSeq: 165 REGISTER >>> Contact: >> IP>:26462>;expires=120;received="sip::26462" >>> Content-Length: 0 >>> <-> >>> --- (8 headers 0 lines) --- >>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961 >>> handle_response_register: Outbound Registration: Expiry for >>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s) >>> Really destroying SIP dialog >>> '005dbc8238e06ac421ef613a3b55e134@127.0.0.1' Method: REGISTER >>> <--- SIP read from UDP:107.6.123.181:5060 ---> >>> <-> >>> Reliably Transmitting (NAT) to 107.6.123.181:5060: >>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport >>> Max-Forwards: 70 >>> From: "Unknown" ;tag=as41ddf4a6 >>> To: >>> Contact: >>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160 >>> CSeq: 102 OPTIONS >>> User-Agent: FPBX-15.0.16.81(16.13.0) >>> Date: Sun, 20 Dec 2020 07:07:07 GMT >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >>> INFO, PUBLISH, MESSAGE >>> Supported: replaces, timer >>> Content-Length: 0 >>> --- >>> <--- SIP read from UDP:107.6.123.181:5060 ---> >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP >>> 192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received=>> OFFICE PUBLIC IP> >>> From: "Unknown" ;tag=as41ddf4a6 >>> To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924 >>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160 >>> CSeq: 102 OPTIONS >>> Content-Length: 0 >>> <-> >>> --- (7 headers 0 lines) --- >>> Really destroying SIP dialog >>> '0b0605df4f4ca7b03402c9fd5a869606@1
Re: [asterisk-users] Playing MP3's in Asterisk
Hi all, Returning to the issue of mp3 support in Asterisk, it seems it is using a build from 1997?! http://svn.digium.com/svn/thirdparty/mp3/trunk/layer3.c I have the same problems as everyone with the mp3 add-on, but now a new one: - "mp3/interface.c: Junk at the beginning of frame 49443304" for ALL mp3 files, even when debug/verbose is off. - Very quiet playback compared to slin, so I have to control the outgoing volume by +7 for mp3 playback. - Now I find it chokes on mp3s which have a png cover art - this is perfectly valid in mp3 id3 But I still need to use ControlPlayback from dialplan and f-agi. I started going down a rabbit-hole of ARI media and Audiosocket etc, for example: https://www.youtube.com/watch?v=tjduXbZZEgI https://github.com/CyCoreSystems/audiosocket These all seem to add a fair amount of complexity for controlling playback - for example: https://wiki.asterisk.org/wiki/display/AST/ARI+and+Media%3A+Part+2+-+Playbacks#ARIandMedia:Part2-Playbacks-Controllingplaybacks And I need to know the current cplaybackoffset at any given time. I was thinking: to keep it simple, is there a way to tell Asterisk to use a different mpg123 (the system one - my currently installed version from 2020!) instead of the Asterisk build? I can live with the "junk" warnings and the quiet playback, but it cannot be crashing on valid id3 content (and I have no control over the remotely hosting files). Thanks for any pointers in the right direction! On Tue, 3 Sept 2019 at 16:20, Dovid Bender wrote: > Ludovic, > > Currently using Asterisk 14.5.0 with > [root@a1-c2-nyc3 ~]# rpm -qa | grep mpg123 > mpg123-1.25.6-1.el7.x86_64 > mpg123-libs-1.25.6-1.el7.x86_64 > mpg123-devel-1.25.6-1.el7.x86_64 > [root@a1-c2-nyc3 ~]# > Should I be using a newer version of mpg123? > > > > On Tue, Sep 3, 2019 at 11:11 AM wrote: > >> Hi ! >> >> Which Asterisk version do you use? >> Asterisk uses the mpg123 library. >> >> Regards, >> >> Ludovic >> >> On Sun, Sep 01, 2019 at 03:27:59PM -0400, Dovid Bender wrote: >> >Hi, >> >Does anyone know what code (if any)Â format_mp3 uses? I am trying to >> play >> >a MP3 (e.g. >> >> -- >> https://drolez.com/blog- Marketing automation and Web dev >> https://chezsandro.com - A cool place in Cape Verde :) >> https://aopensource.com- Android Open Source Portal >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
